Diy digital crossover

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TAS5706: single chip, 2 speakers. no aditional clock.
TAS5518: PWM controller, up to 8 speakers, requires separated Driver IC, external 50MHz.

Biquad parameter: https://sites.google.com/site/koonaudioprojects/tas5518-full-digital
on the bottom you can find C source code to calculate LPF/HPF Biquad parameter.

I understand this thread is for Active Speaker, if the speaker has Digital In and Analog In, the speaker will have CS8416 for SPDIF receiver / ADC for Analog In / switch and data selector like 74HC157.
 
TAS5706: single chip, 2 speakers. no aditional clock.
TAS5518: PWM controller, up to 8 speakers, requires separated Driver IC, external 50MHz.

Biquad parameter: https://sites.google.com/site/koonaudioprojects/tas5518-full-digital
on the bottom you can find C source code to calculate LPF/HPF Biquad parameter.

I understand this thread is for Active Speaker, if the speaker has Digital In and Analog In, the speaker will have CS8416 for SPDIF receiver / ADC for Analog In / switch and data selector like 74HC157.

Good stuff! Is your program also compatible with TAS3108 (can program it using I2C), I think it's got the same DSP core inside?

I'd like to design a PCB for the Crystal S/PDIF receiver (CS8416, or rather a sample rate converter chip to avoid S/PDIF jitter to go all the way to the converters) and TAS3108 using Asus Xonar H6 (http://www.asus.fi/Multimedia/Audio_Cards/Xonar_H6/) as analog output board.
 
Last edited:
mhelin,
TAS3108 looks like generic DSP, combined with TI's software?
"The devices use an efficient, custom, multi-instruction programming environment optimized for digital audio processing algorithms."
PurePath Studio Graphical Development Environment (GDE)

I saw TAS3103, it has configurable Biquad filters.
Setting procedure should be very different from TAS5706/TAS5518, but calculation for Biquad will be the same issue.
 
Dear All,

It seems you are all talking hardware now.

The real problem lies in proper programming software for whatever platform you are using. Standard filter curves will not produce acceptable results, since the invidual SPL characteristics of the drivers-in-box have to be taken into consideration, and therefore must be measured first. Then the measurement results must be imported into the software.

Finally, filter+acoustic raw driver output must be calculated such that a chosen target. e.g. 4th order LR output is produced acoustically, not electrically.

Only a handfull of software packages can do both DSP programming and filter optimization, but SoundEasy, LSP CAD 6.1.and LEAP can. To my best of knowledge, the standard Behringer DCX 2496 software cannot do this, thus producing acoustically undesired results.

Eelco
 
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